My PBX is behind NAT and I have trunk connect to GoIP device (for callout using Mobile number)
I use softphone (pjproject, linphone) to make call to my phone number, I can hear the sound normally.
But when I use WebSocket (sipml5, sip.js) to make a call, I don’t hear any sound and the call automatically hangs up after about 15-20 seconds.
It’s weird that bundle=no is being set in pjsip.endpoint.conf, but I’m guessing that’s not the root cause of your trouble.
I don’t see anything other than host candidates in the SDP from the PBX to the webrtc client. Since the PBX is behind NAT, it’s probably not a bad idea to setup a stun (or even a turn server) in rtp.conf.
If that doesn’t resolve it, you’re going to probably need to look deeper here - so do an rtp set debug on on the Asterisk CLI when the webrtc call is up and verify that you have RTP packets being received and sent to the endpoint. Maybe some PCAP from both sides to verify it as well.
It’s weird that bundle=no is being set in pjsip.endpoint.conf, but I’m guessing that’s not the root cause of your trouble.
==> I don’t know what this setting is for, but the pjsip.endpoint.conf file is generated from FreePBX (ver: 15.0.23) and I did not edit anything in that file.
On the other hand, I see in bundle=no in the sip table of asterisk DB
I don’t see anything other than host candidates in the SDP from the PBX to the webrtc client. Since the PBX is behind NAT, it’s probably not a bad idea to setup a stun (or even a turn server) in rtp.conf.
==> I added STUN server to RTP’s settings and my call heard the sound.
